The purpose of this study was to determine the effects of on-screen video instructor gender and expressivity upon the competency based performance of adults in the learning of basic computer skills from a series of locally produced instructional videotapes. These tapes were typical of the type of instructional videos used by corporations, government, and schools to train adults. A 2 x 3 factorial design was used to analyze the variables effects on learning. The independent variables were instructor gender and instructor expressivity. Mediating variables were student gender, ethnicity (as measured by first language), reading level, and attitude towards computers. The dependent variable was student learning as measured by a 20 question multiple choice exam. The subjects (n=120) used in the study were under employed or unemployed adults. Sixty-nine percent were considered to be limited speakers of English. Six videotapes were produced. These tapes were identical in every way except that in three tapes a male actor taught in respectively low, average, and high expressive manners. In each of the other three tapes a female acted in a low, average and high expressive manners. Each subject viewed one of the tapes, in groups of eight to ten students, and was immediately tested on the material presented in their tape. Students were also asked to rate the instructor they saw on a five item scale of expressivity. The findings showed no significant main effects for either on-screen instructor gender or expressivity. A significant interaction was found between on-screen instructor gender and expressivity, as measured by the students. Students viewing the low expressive male and the high expressive female scored higher on the exam than students viewing either the high expressive male or the low expressive female. Performance on the test was correlated with student reading level, computer experience, the time it took to take the exam, and student ethnicity (as measured by first language spoken). Additionally, a significant interaction was found between student gender and instructor gender. Students viewing instructors of the opposite gender scored higher than those viewing instructors of the same gender. The implications of this study suggest that subject matter and student population be considered in both the design of instructional television programs and in the casting and the directing of talent for these programs. / Ed. D.

Det här arbetet syftar till att genom design av en telefonapplikation undersöka hur ljud kan förstärkas av den omgivande miljön. Arbetet visar på hur miljöförstärkt ljud skiljer sig från ljudförstärkt miljö samt vilka begränsningar det innebär att använda en vanlig smartphone som plattform.  Ett långsiktigt mål är att genom en produkt som låter oetablerade ljudproducenter distribuera sina alster i den offentliga miljön främja fri kultur och kreativitet. Arbetet har följt principen för RtD (research through design), där designprocessen utgör det fall som studeras och dokumentationen av den utgör den bas av data som analyseras och leder till slutsatser kring ämnet. Designprocessen i detta arbete innebär en mindre omfattande användarstudie i enkätform för att identifiera målgrupp, krav och mål. Följd av en andra studie där en prototyp framtagen med grund i resultaten från användarstudien utvärderas av deltagare genom observation, intervju och reaktionskort.  De största utmaningarna för designen grundar sig i att ta fram en applikation som inte stör upplevelsen, men samtidigt kan instruera användaren i hur man använder applikationen. Utifrån arbetet dras slutsatsen att användarinstruktioner, såväl visuella som auditiva, löper större risk att störa upplevelsen i miljöförstärkt ljud än i ljudförstärkt miljö. Vidare konstateras att den största bristen i en lösning som bygger på standardutrustning i jämförelse med en lösning som nyttjar dedikerad utrustning utgörs av begränsningar när det kommer till att ange riktning med hjälp av ljudets rumslighet.

The purpose of this project was to design, create, and implement a pilot for an adapted physical education (APE), DVD-Video teaching aid. This creative project has resulted in a DVD-Video that can be used as an instructional tool and resource for physical educators. It follows the teaching methods and skills from the wheelchair basketball chapter in Davis (2002), featuring strategies and ideas that can aid in the successful teaching of physical education (basketball) to individuals of all abilities.The use of interactive DVD-Video in a learning environment is seen as an innovative and exciting delivery method for instructional purposes. It is proposed that the majority of learners would enjoy using, and acquire a very significant learning experience from using, interactive DVD-Video instruction. Therefore, it is anticipated that physical educators would get a meaningful APE training experience from an interactive instructional DVD-Video such as the one piloted for this creative project. / School of Physical Education, Sport, and Exercise Science

Visualization, processing and editing of audio, directly on a time-frequency surface, is the scope of this thesis. More precisely the scalogram produced by a Gabor Wavelet transform is used, which is a powerful alternative to traditional techinques where the wave form is the main visual aid and editting is performed by parametric filters. Reconstruction properties, scalogram design and enhancements as well audio manipulation algorithms are investigated for this audio representation.The scalogram is designed to allow a flexible choice of time-frequency ratio, while maintaining high quality reconstruction. For this mean, the Loglet is used, which is observed to be the most suitable filter choice.  Re-assignmentare tested, and a novel weighting function using partial derivatives of phase is proposed.  An audio interpolation procedure is developed and shown to perform well in listening tests.The feasibility to use the transform coefficients directly for various purposes is investigated. It is concluded that Pitch shifts are hard to describe in the framework while noise thresh holding works well. A downsampling scheme is suggested that saves on operations and memory consumption as well as it speeds up real world implementations significantly. Finally, a Scalogram 'compression' procedure is developed, allowing the caching of an approximate scalogram.

This qualitative research aims to present students' beliefs about the role that culture plays in language learning and to understand their responses towards the use of authentic multi-media materials. Based on the results of the study, this dissertation describes the impact that authentic multi-media materials had on students' cultural knowledge and their overall language learning experience. The implications discussed in this study are based on students' responses and are relevant to foreign language instructors who are interested in the areas of culture teaching, authentic multi-media materials, students' perspectives, and autonomous learning. In particular, the dissertation presents a link between authentic multi-media materials and motivation based on Ryan and Deci's Self-Determination Theory (2000). The results found in this study suggest that authentic multi-media materials are resources that could be utilized in the early stages of instruction in order to capitalize on students' inherent interest in the Italian culture, which in turn serves to motivate students to persist in their language study. / text

Modified discrete cosine transform (MDCT) derived from DCT IV has emerged as the most suitable choice for transform domain audio coding applications due to its time domain alias cancellation property and de-correlation capability. In the present research work, we focus on MDCT domain analysis of audio signals for compression and other applications. We have derived algorithms for linear filtering in DCT IV and DST IV domains for symmetric and non-symmetric filter impulse responses. These results are also extended to MDCT and MDST domains which have the special property of time domain alias cancellation. We also derive filtering algorithms for the DCT II and DCT III domains. Comparison with other methods in the literature shows that, the new algorithm developed is computationally MAC efficient. These results are useful for MDCT domain audio processing such as reverb synthesis, without having to reconstruct the time domain signal and then perform the necessary filtering operations. In audio coding, the psychoacoustic model plays a crucial role and is used to estimate the masking thresholds for adaptive bit-allocation. Transparent quality audio coding is possible if the quantization noise is kept below the masking threshold for each frame. In the existing methods, the masking threshold is calculated using the DFT of the signal frame separately for MDCT domain adaptive quantization. We have extended the spectral integration based psychoacoustic model proposed for sinusoidal modeling of audio signals to the MDCT domain. This has been possible because of the detailed analysis of the relation between DFT and MDCT; we interpret the MDCT coefficients as co-sinusoids and then apply the sinusoidal masking model. The validity of the masking threshold so derived is verified through listening tests as well as objective measures. Parametric coding techniques are used for low bit rate encoding of multi-channel audio such as 5.1 format surround audio. In these techniques, the surround channels are synthesized at the receiver using the analysis parameters of the parametric model. We develop algorithms for MDCT domain analysis and synthesis of reverberation. Integrating these ideas, a parametric audio coder is developed in the MDCT domain. For the parameter estimation, we use a novel analysis by synthesis scheme in the MDCT domain which results in better modeling of the spatial audio. The resulting parametric stereo coder is able to synthesize acceptable quality stereo audio from the mono audio channel and a side information of approximately 11 kbps. Further, an experimental audio coder is developed in the MDCT domain incorporating the new psychoacoustic model and the parametric model.

Musical timbre transfer is the task of re-rendering the musical content of a given source using the rendering style of a target sound. The source keeps its musical content, e.g., pitch, microtiming, orchestration, and syncopation. I specifically focus on the task of transferring the style of percussive patterns extracted from polyphonic audio using a MelGAN-VC model [57] by training acoustic properties for each genre. Evaluating audio style transfer is challenging and typically requires user studies. An analytical methodology based on supervised and unsupervised learning including visualization for evaluating musical timbre transfer is proposed. The proposed methodology is used to evaluate the MelGAN-VC model for musical timbre transfer of drum tracks. The method uses audio features to analyze results of the timbre transfer based on classification probability from Random Forest classifier. And K-means algorithm can classify unlabeled instances using audio features and style-transformed results are visualized by t-SNE dimensionality reduction technique, which is helpful for interpreting relations between musical genres and comparing results from the Random Forest classifier. / Graduate

The Web Based Audio Editor thesis deals with the requirements specification, selection of technologies and with the implementation itself. The thesis uses modern approaches of HTML5 standards and is divided into the client and the server parts. Both the web application and the server are implemented in JavaScript. The Web based audio editor allows basic editing features like cut, shifting, deleting. All this in multiple audio tracks. The application runs in recent versions of the most widely used web browsers.

The dissertation thesis focuses on objective assessment and reduction of disturbing background noise in a musical signal. In this work, a new algorithm for the assessment of background noise audibility is proposed. The listening tests performed show that this new algorithm better predicts the background noise audibility than the existing algorithms do. An advantage of this new algorithm is the fact that it can be used even in the case of a general audio signal and not only musical signal, i.e. in the case when the audibility of one sound on the background of another sound is assessed. The existing algorithms often fail in this case. The next part of the dissertation thesis deals with an adaptive segmentation scheme for the segmentation of long-term musical signals into short segments of different lengths. A new adaptive segmentation scheme is then introduced here. It has been shown that this new adaptive segmentation scheme significantly improves the subjectively perceived quality of the musical signal from the output of noise reduction systems which use this new adaptive segmentation scheme. The quality improvement is better than that achieved by other segmentation schemes tested.

No description available.